In this page you find a review of articles dedicated to mixing and post production.
27 August 2018
12 essential pro mastering tips
The art of mastering is shrouded in myth and mystique, but with these 12 no-nonsense, practical tips, we aim to clear the conceptual fog and help you get your finished tracks sounding better than ever.
1. Hands-off mixing
Don’t expect mix problems to be solved at the mastering stage! EQ clashes, dynamic issues and other errors are all best addressed from within the mix project. If you’re applying drastic amounts of processing, revisit your mix – or if you’re mastering for someone else, explain the issues and see if they can remix.
2. Keep it fresh
Don’t over-listen to a track! Pro mastering engineer John Paul Braddock explains: “What I don’t want to do is to listen all the way through the track for six minutes, because as soon as I’ve done that, I’ve got used to how it sounds, rather than being objective. It’s crucial that we don’t spend too much time listening to the music. This might sound counter-intuitive, but we’re not mixing it any more. We’re not trying to listen to the detail; we’re trying to get an overview – to sample the overall tone of the song.”
3. Consider it
Once you’re ready to master a track, don’t just dive in and start processing. The aim is to gently improve, not ‘mix’. Take a more considered approach. Briefly compare the mix to a reference track at equal level, plan exactly what correction or enhancement the mix needs, try it out, re-level, then evaluate.
4. On the knobs
Type in parameter values and use stepped plugins (with fixed 0.5-1dB ‘notched’ controls) where possible. It’s easy to just crank up a knob, but typing in values makes you think about what you’re entering. Stick to 0.5/1dB steps at a time, as half a dB will make a significant difference when mastering.
5. Does it cancel out?
Use ‘sum difference’ testing (also known as a ‘null test’) to hear if a plugin is ‘passive’. Get to know which plugins add gain boosts or frequency changes in their default state. John Paul Braddock again: “Many plugins will actually apply a tonal or level change even before any settings are dialled in. I’ve noticed that, after analysing several types of plugins, you’ll load up a plugin with ‘no processing’, but the actual output might be louder. Perhaps those plugin manufacturers know what we now know – that ‘louder sounds better’ – or maybe that’s just a side effect of the plugin’s design. The important thing is that you analyse the tools you’re using, and don’t make assumptions. Be critically aware of your own tools.”
6. Reference with equality
Remember to compare your final processed master with the unprocessed session mix – at equal level – to see if you’ve actually achieved the outcome you intended. If not, don’t be afraid to start again from scratch.
7. Staged limiting
Several gentle stages of limiting or compression can help take the load off one single plugin. For example, three limiters with gain reduction of 1dB might sound more natural than a single 3dB limiting stage. It depends on the plugins used, so give it a try, and listen objectively.
8. Mid/side DIY
A plugin with an unlinked left/right mode can also be used to process in mid/side. Simply load Voxengo’s free MSED on the channel and set it to Encode. Now load your plugin after MSED and unlink the left and right channels. Place a second MSED last in the chain, and set it to Decode. The left side of your plugin now processes the mid (mono) part of your signal, and the right affects the side (stereo).
9. Make it up
For transparency, try to use as few EQ or excitement stages as possible. So, if a track has too much bass and not enough treble, try using a single broad shelf to cut bass, then re-level by increasing the EQ’s makeup gain. This will shift the track’s weight towards the treble more naturally than two EQ bands.
10. More than average
Regular downwards, full-band compression can clamp down on peaks and transient detail, ruining dynamics if not applied carefully. If your track needs extra average weight, consider blending it in through the use of parallel compression – you can bring up the average level of your track while keeping the detail intact.
11. Compress gently
A touch of downward compression can pull (or ‘gel’) the overall mix together, but keep attack times slow so you gently clamp down on the mix’s sustain and not the transients. A low ratio and around 1-2dB of gain reduction should be all that’s necessary.
12. Limit last
Many think of limiting and loudness as the main staples of mastering, but this attitude often leads to amateur results, flattened mixes and distortion. Final peak limiting should only be tackled when a track’s overall tonal, dynamic and stereo balance are in order. So leave limiting till last!
14 July 2018
The 10 Best Studio Monitor Speakers – Essential Buyers Guide 2018
After your computer and DAW software, studio monitors are arguably the next most important component in any music production environment (with the possible exception of the audio interface.
When levelling up your music production, the importance of a decent set of monitors cannot be underestimated, since their role in accurately transmitting the sound from the outputs of your interface across the room to your ears makes them as vital a link in the chain as the interface itself.
A good set of monitors should be the audio equivalent of a mirror – what you hear should be a true representation of what’s actually there. The more accurate the response, the more likely it is that, if it sounds good to your ears, itwill sound good to everybody else as well. The question is, what are you looking for in the ideal project / home studio monitor if your main output is electronic or computer-based music?
Monitor selection is a very personal thing – what works for one person may not work at all for another. You’ll be looking for something with a compact footprint that can be positioned fairly close to a wall without compromising the bass response, a deep bass extension for those kicks and sub basses, coupled with a non-flattering, even response across the mids and crisp, detailed highs. You want a monitor to accurately reflect the changes you’re making to your mix, the ultimate aim being to produce an end result that sounds good when played back in any environment, from your car, to the club, to the Bluetooth speaker you’re playing your iPhone through at your mate’s barbecue.
Monitors that exhibit a so-called ‘smiley curve’ response, with exaggerated bass and top-end responses designed to flatter the sound, may sound good while you’re working on them, but the end result is likely to be a lacklustre mix lacking in lows and highs. After all, while you’re working on your project, it’s what you hear with your own ears that steers the decisions you make when it comes to cutting or boosting certain frequencies and levels. So if your monitors are colouring the sound and making it sound more punchy, bottom-heavy and sparkly than it really is, you’re going to end up disappointed when you play your mix elsewhere.
“A good set of monitors should be the audio equivalent of a mirror – what you hear should be a true representation of what’s actually there.”
The difference, then, between studio monitors and the consumer speakers that might come with your hi-fi or computer, is that studio monitors are designed to reproduce audio frequencies as accurately as possible across the audible frequency spectrum, ideally at any volume level. While a totally flat response across the frequency spectrum is the Holy Grail of monitor manufacturers, it’s an inescapable fact that all monitors will colour the sound in some way. It can take a while to get used to the particular characteristics and sound of one set of monitors over another. The longer you use them, the more you get to know the quirks in their response curves and tailor your production decisions accordingly to ensure the best end result.
Achieving such a flat response can come at a cost, however, as it tends to take quality materials, high-end electronic components and well-executed design and engineering to pull this off, which is why a decent pair of monitors can be a bit on the pricey side. However, there are still plenty of options available that are capable of delivering more than adequate performance for those on a tighter budget.
Speakers corner – the evolution of the modern monitor
Before the advent of tape recorders, most commercial recordings were cut live to a master disk, often in a single take and with no editing. Because of this, there was little or no real need for accurate monitoring, as playback scenarios were few and far between. With the advent of tape recording, multitracking, overdubbing and mixing however, there came a heightened requirement for being able to focus in on the detail of what was being recorded and played back after the event.
Up until the late 1960’s, studio monitoring systems tended to consist solely of large, wall-mounted units with huge cones for the low frequencies and concentrically-mounted midrange cones or horns. Each component was fed by a crossover unit that split the incoming signal into the required frequency bands, usually at around 1000-1200 Hertz, so that the low frequencies were sent to the bass cone and everything else to the midrange unit.
By the 1970’s, however, studio technology had progressed to the point where nearfield monitoring was becoming more widely used, driven in part by the BBC’s need to monitor in the vans they were using for outside broadcasts at the time. The BBC’s LS3/5A and JBL’s 4310 compact nearfield monitors brought about a revolution in studio monitoring. These units were compact enough to be placed on the top of the mixing desk and could be listened to at much closer distances than their wall-mounted cousins. They also allowed the engineer to focus more on the sound coming directly from the speakers, rather than that being reflected off the walls and ceiling. So began a trend that has lasted up to the present day, the technology having progressed over the intervening years to the point where most ‘nearfield’ studio monitors are compact and affordable enough to bring professional-level performance within reach of the budget-conscious home recordist / producer.
Key features to consider
As always when purchasing any piece of recording gear, there are several factors that can influence where your money goes. Here are just a few of them…
Active vs passive
Passive monitors by definition need to be connected to a compatible power amp in order to work – a notable example is the famous Yamaha NS10, the iconic, white-coned nearfield monitor that dominated the professional studio market in the 80’s and 90’s, whose perfect partner was the famous Quad 405 power amp. The one main advantage of this approach was the versatility it afforded to customise your system – changing the amp could have a major effect, either detrimental or beneficial, on the sound of your monitoring setup. It also meant that the speaker units themselves were lighter and therefore easy to set up and move around.
These days, thanks to the progress made in the miniaturisation of electrical circuits and components required in loudspeaker design in general, most nearfield monitors are of the active type; i.e. they don’t require external amplification because all the necessary amps and crossovers required to drive the conesare built into the speaker casing along with the cones themselves. This is more convenient not just because the units are self-contained, but also because you don’t need to shell out an additional arm and a leg for a compatible power amp. Most active monitors also offer some form of adjustable room response compensation features, achieved by the use of internal digital signal processing chips (DSP) that can be a bonus when setting up and adjusting to your unique monitoring environment. You will, of course, require a power lead (and a mains socket of course) for each speaker, but that’s not really an issue in most modern studio environments. So because the vast majority of monitors on the market today are active designs, unless you specifically want to go the passive route, this is largely a decision that has already been sorted for you by the manufacturers.
Reflex / Transmission line
Smaller speakers are always going to struggle more to accurately reproduce bass frequencies than larger units, simply because of the wavelengths of low-frequency soundwaves. One way in which designers have sought to get around this is by building bass ‘ports’ into their speaker casings, an approach known as reflex design. These are specially designed ‘tunnels’ within the casing that exit via a hole – a bit like an exhaust pipe – somewhere in the casing of the speaker unit. If the bass ports are located on the rear of the cabinet however, this can cause issues with positioning of the monitors close to a wall.
A variation is the transmission line design, which gets around this problem by incorporating a sort of long, folded tunnel structure, lined with absorbent acoustic material, into the innards of the cabinet to handle the bass response, often terminating in the front of the speaker. This can significantly reduce low frequency distortion and deliver greater bass extension and loudness than a ported or sealed design of a similar size, meaning that you don’t have to crank the volume to elicit the correct bass response – the response is balanced whatever the volume level.
A good set of monitors with a so-called flat frequency response should theoretically give you the ability to reveal detail and make mix decisions such as subtle changes to EQ and compression more easily. The spec sheets will usually quote a range of frequencies that the speakers are able to reproduce, which will usually exceed those frequencies audible by the human ear, normally around 20Hz – 20kHz. Many monitors also offer options for tuning the frequency response of the monitors to match the acoustics of your studio space – this is done by the addition of digital signal processing (DSP) circuitry, which in turn requires internal analogue-digital-analogue (ADA) converters, the quality of which can have a big effect on the overall sound.
Woofers and tweeters
One thing that all the monitors on our list have in common is that they all have multiple drivers. In other words, each speaker unit has at least two cones, one large one (or ‘woofer’) to handle the bass and lower-mid frequencies, and a small one (or ‘tweeter’) to take care of the upper mids and highs. In an active monitor, the incoming audio signal is split into two frequency bands – low and high – by a component called a ‘crossover’. The point at which the split occurs is known as the crossover frequency, and this varies greatly from monitor to monitor, anywhere from between 800Hz-2kHz. The low-frequency signal is then sent to the woofer and the high-frequency signal to the tweeter, each via its own power amp. The fact that each driver has its own amplifier is often what contributes to the surprising weight of an active monitor!
Mix sessions can last for hours at a a time, so one very important thing to consider when choosing studio monitors is so-called ‘ear fatigue’, which essentially translates into how long you’re able to listen to them at a decent working volume before your ears start to get tired. A revealing sound is all well and good, but if this results in a sound that’s too harsh and unforgiving in the midrange, say, your monitors will be challenging to listen to for any great length of time. Because everyone’s ears are different though, one man’s sonic silk purse may be another’s sow’s ear, so the fatigue factor is something very subjective that’s going to be very hard to gauge without a thorough test of the speakers you have in mind. So if at all possible, try to get hold of a pair to ‘test drive’ for a while, so that any long-term compatibility issues between the speakers and your ears can be assessed. You’ll find it pays dividends in the long run, as you’re more likely to end up with a monitoring system that’s right for you, in that it’s effective and comfortable to work with for long periods.
Sub or no sub?
While perfectly capable as functioning as a stereo pair with good performance across the entire frequency spectrum, many monitors currently on the market are available as so-called 2.1 systems, consisting of two nearfield, desk-mounted stereo speakers to handle the upper and mid-frequency ranges and a separate, single sub-woofer, usually placed somewhere below the desk, to handle low frequencies. Because the sub is usually a large unit containing a large diameter cone, it’s often able to extend the low-frequency response of the system down to around 20Hz or so – the kind of bass you feel in your chest rather than hear. While this may sound great and reveal a lot more low-end detail than a solitary pair of desk-mounted nearfields, in small rooms the extended low-frequencies can create noticeable peaks and troughs in the bass response. For that reason, we’ve predominantly focussed our list on stereo systems, but highlighted whether or not a compatible sub-woofer is available from the same manufacturer.
10 Best Studio Monitors – Buyers Guide List 2017
So now onto the list of what we consider to be ten of the best studio monitors on the market today. Like all of the lists we feature on GTPS, we don’t believe it’s particularly useful to say that one device we’ve included is categorically better than another, and neither is this a categorical list of the ten best monitors available. Monitors are a very personal choice, and in this roundup of some of the market leaders we’re simply highlighting some of the best options currently available for various budgets and studio setups – everyone will, of course, always have their own personal favourites.
1. Focal Shape 65
Focal’s Shape range of active monitors replaces their acclaimed CMS series, and is available in three sizes – the 40, 50 and 65, in order from smallest to largest. The largest of the range, the 65 is designed with a sideways-facing, 6.5-inch passive radiator cone – in essence an unpowered speaker cone that reacts to the sound pressure inside the cabinet. This dismisses the need for a bass port, the idea being that this makes it easier to position the monitors with their backs next to a wall, a commonly-required configuration compromise in a lot of small studios. There’s also low tweeter directivity, meaning that it’s not so essential to have the speakers placed at one particular optimal angle for best results. Sound-wise, with a variable crossover set at 160Hz and adjustable low and high frequency settings to optimise the response to their surroundings, the Shape 65 offers great performance for the price, and as a bonus the cabinets include built-in threaded holes so the speakers can be wall or ceiling-mounted using an optional accessory mounting kit.
Focal Shape 65 – Specs:
Design: 2-way active, with side-mounted, 6.5″ double passive radiator
Woofer: 6.5-inch Flax sandwich cone
Tweeter: 1-inch ‘M’ profile Aluminum-Magnesium dome
Frequency response: 40Hz – 35kHz
Wattage: MF/LF 80W, class AB, HF 25W, class AB
Dimensions: 14 x 8.6 x 11.2″ (355 x 218 x 285mm)
Weight: 18.7lb (8.5kg)
Compatible Sub? No
2. Yamaha HS7
Yamaha all but cornered the nearfield market back in the 80’s and 90’s with its legendary NS10M monitors. In a matt black wooden casing with a distinctive white bass cone, they came to be the monitors of choice for a huge percentage of studios worldwide, thanks to their uncompromisingly accurate response. The NS10M was discontinued by Yamaha in 2001, but its DNA lives on in the HS series, of which the HS7 is the mid-sized variant. Sticking with that white cone, the HS7 builds on its heritage, adding an active reflex design with a rear-facing bass port, and includes a pair of switches to boost or attenuate high and low frequencies by a dB or two, so as to tailor the response to the room you’re using them in. At a very reasonable price point designed to catch the attention of home studio owners (hence the ‘HS’ moniker) who like the idea of some white-coned Yammies on their desk, the HS7’s deliver a lot of bang for the buck.
Yamaha HS7 – Specs:
Design: 2-way active, bass reflex (rear ported)
Woofer: 6.5-inch cone
Tweeter: 1-inch dome
Frequency response: 43Hz – 30kHz
Wattage: LF 60W, HF 35W
Dimensions: 8.3 x 13.1 x 11.2″ (210 x 332 x 284 mm)
Weight: 18lb (8.2kg)
Compatible Sub? Yes – Yamaha HS8S
3. Dynaudio Acoustics LYD 5
Dynaudio Acoustics are responsible for some legendary high-end speaker designs, including the acclaimed BM15A. For those looking to acquire a taste of Dynaudio quality in a more affordable package, there’s the LYD series, made up of the LYD 5 and its bigger brother, the LYD 8. Both available either in a resplendent white or standard black finish, the LYD 5 (the 5 denotes its 5-inch woofer cone size) packs quite a punch, belying its diminutive size with a beefy sound achieved in part by a rear-facing bass port. Any issues this may cause by having the speakers positioned against a wall can be offset by the ‘Position’ mode, while the Sound Balance control is an attempt to compensate for overly bright or dead rooms. There’s also a Bass Extension control to tailor the low end to your working environment. All in all then, a smart-looking, versatile monitor from a pedigree brand at a decent price – what’s not to like?
Dynaudio Acoustics LYD 5 – Specs:
Design: 2-way active
Woofer: 5-inch MSP cone
Tweeter: 28mm soft dome
Frequency response: 50Hz – 22kHz
Wattage: LF 50W, HF 50W
Dimensions: (170 x 260 x 211mm)
Weight: 12.5lb (5.7kg)
Compatible Sub? No
4. MunroSonic Egg 150 Monitoring System
Up there as contenders for most unusual speaker design, the MunroSonic Egg range of monitoring systems lives up to its name, in that it consists of a pair of uniquely egg-shaped passive speakers driven by a dedicated external analogue power amp that comes as part of the package, along with two high-quality, 3m cables to connect it all up. There is, naturally, a good reason for this departure from the norm – the company’s literature states that ‘the traditional wooden box has been replaced by a scientifically proven, curved enclosure that virtually eliminates diffraction and resonant effects that distort and smear the original sound’.
Combined with the free-standing amp is a control unit that features source-select inputs, active analogue crossovers, LF and HF trim pot equalisation for room and location set-up compensation and a re-defined mid-band control to emulate the mid-range response of both Hi-Fi and NS10 type speakers. The egg-shaped driver units sit on purpose built ‘nests’ that allow flexible positioning in all directions, aided by blue guide LED’s above each tweeter to help you locate the ‘sweet spot’ when setting up. The nests also allow sufficient room for the downward-facing bass port to expel the necessary air.
MunroSonic Egg 150 Monitoring System – Specs:
Design: 2-way passve, with downward-facing bass port and external analogue power amp
Woofer: 165mm Polypropylene Cone
Tweeter: 25mm dome
Frequency response: 45Hz – 20kHz
Wattage: MF/LF 50W HF 50W
Compatible Sub? No
5. Eve Audio SC204
The smallest monitor in Eve Audio’s product line, the SC204 combines a 4-inch honeyomb-structured driver with an AMT (Air Motion Transformer) ribbon tweeter of their own proprietary design that’s used throughout the whole Eve Audio range. RCA and XLR analogue inputs feed directly into Burr-Brown digital converters, and the signal is processed digitally before reaching the PWM Class-D digital amps that feed the drivers. Sophisticated stuff.
A unique addition is the front-mounted volume control – the sort of thing you’d normally expect to find on budget desktop computer monitors, except this knob controls a host of other features, including standby mode, filter selections, speaker symmetries, phase tweakings and even LED display intensity.
There’s a large rear rectangular port with smooth edges to extend the low-end response and minimise bass distortion, while the ribbon tweeter provides a crisp and detailed high end that’s bright without being harsh. All in all, these diminutive and affordable speakers punch way above their weight.
Eve Audio SC204 – Specs:
Design: 2-way active
Woofer: 4-inch cone
Tweeter: AMT RS1
Frequency response: 64Hz – 21kHz
Wattage: MF/LF 50W, HF 50W
Dimensions: 5.7 x 9.0 x 7.7″ (145 x 230 x 195mm)
Weight: 8.4lb (3.8kg)
Compatible Sub? Yes – Eve Audio TS107
6. PMC twotwo 6
British manufacturer PMC aren’t just known for the excellent quality of their speakers, but also for the transmission line principle that lies at the core of their design philosophy. Their version is named ATL (Advanced Transmission Line) and, in the twotwo range, places the bass cone near one end of a long tunnel within the casing, lined with material that absorbs the high and low-mid frequencies radiating from the rear of the cone. The lowest frequencies emerge from the end of the tunnel via a port at the front that effectively acts as a second bass driver.
The twotwo range consists of three models – numbered 5, 6 and 8 after the size of the bass driver in each – and what sets them apart from their predecessors is the addition of DSP-based EQ, crossover and driver response adjustment, making them the only PMC nearfield monitors to feature EQ compensation apart from the range-topping AML series.
Throw in an LCD screen on the back to let you see what’s happening while you adjust the EQ settings, digital and analogue inputs and the ability to link monitors together with a Cat 5 ethernet cable that supplies duplicate EQ settings and audio to a second monitor and you have a class-leading package that can’t be ignored.
PMC twotwo 6 – Specs:
Design: 2-way active
Woofer: 6.5-inch cone
Tweeter: 27mm soft dome
Frequency response: 40Hz – 25kHz
Wattage: MF/LF 150W, class D, HF 50W, class D
Dimensions: 16” x 7.6 x 14.3” (406 x 194 x 364)
Weight: 18.5lb (8.4kg)
Compatible Sub? Yes – twotwo sub1 & twotwo sub2
7. Adam A7x
Another brand that favours the ribbon tweeter design, Adam Audio’s version is known as the X-ART (short for eXtended Accelerating Ribbon Technology), so-called because of a claimed flat frequency response that extends up to an amazing 50kHz. Why would you want to go that far beyond the range of human hearing you might ask? Good question, but the idea behind the design is to achieve detailed, uncompressed highs and upper mids without being tiring over long listening periods, and it has to be said that the A7x reproduces transients with incredible clarity, creating an extremely precise sound. Universally accepted as one of the best monitors on the market, the A7x improves on the original A7 model in almost every aspect, from that updated tweeter and new amps to a redesigned bass and midrange driver. There’s an additional bass port on the front, and round the back you have controls for tweeter level and high and low shelf filters.
Adam A7x – Specs:
Design: 2-way active
Woofer: 7-inch Carbon/Rohacell/Glass Fibre cone
Tweeter: X-ART (Equivalent diameter 2”)
Frequency response: 42Hz – 50kHz
Wattage: MF/LF 100W PWM, HF 50W class AB
Dimensions: 13.5 x 8 x 11″ (337 x 201 x 280mm)
Weight: 20.3 lb (9,2 kg)
Compatible Sub? Adam Audio Sub 10 Mk 2
8. ATC SCM25A
The SCM25A is ATC’s first ever compact three-way active studio monitor, but their previously available large / midfield designs have garnered accolades from every corner of the the music-making world, building a reputation for precise and detailed reproduction across the whole frequency spectrum at any volume level. The SCM25A continues the family tradition, and at a whopping 30kg per cabinet, you know you’re dealing with a quality product as soon as you (and three burly mates) lift it from the box!
Design-wise, it’s one of the few on the market to offer a choice between two different approaches to loudspeaker cabinet design. There’s a large port vent on the side panel adjacent to the woofer that can be plugged with a dense foam bung, transforming the speaker from a reflex design into a sealed cabinet design if needed.
With a clean and natural bottom end, the mid-range clarity shines through thanks to that renowned ATC soft-dome mid-range driver, and the highs are honest and detailed, allowing for easy analysis and precise adjustment of mixes without being overly fatiguing or clinical. There’s no escaping the fact that the SCM25A is one of the more expensive monitors on the market for its size, but the accuracy and quality of its performance really does justify the extra expense.
ATC SCM25A – Specs:
Design: 3-way active, with side-mounted bass port
Woofer: hand-built 7˝/164mm short-coil carbon-paper cone
Mid: hand-built 3˝/75mm soft-dome
Tweeter: 25mm neodymium soft-dome
Frequency response: 47Hz – 22kHz
Wattage: LF 150W, MF 60W, HF 25W
Dimensions: 10.4 x 16.9 x 114.5” (264 x 430x 369mm)
Weight: 66lb (30kg)
Compatible Sub? No
9. Barefoot Sound MicroMain 45
For those lucky souls where budget is no object, Barefoot Sound’s range of high-end monitors have to be high on the list of considerations. Barefoot are kind of the monitor equivalent of the Rolls Royce, but nestling down near the more affordable end of their product spectrum (although still not exactly cheap at £6k a pair) the MicroMain 45’s are a stripped-down version of the flagship MiniMain12’s. These three-way active monitors are probably equally suited to midfield and nearfield use owing to their size – and the 8-inch bass driver, twin 2.5-inch midrange cones and 1-inch ring radiator tweeter certainly get the job done.
Barefoot’s approach to monitor speaker design has always been to deliver outstanding sound quality, wide dynamic range and bass extension at every stage of the audio production process, and the MM45 sums up that philosophy in a nutshell. Now though, Barefoot speakers offer a range of preset EQ options in the shape of MEME voice emulation technology, meaning that, with just the flick of a switch, you can get your monitors to emulate the sound of sweetened ‘Hi-Fi’ speakers or, with the ‘Old School’ setting, the classic Yamaha NS10M. Amazing.
Barefoot Sound MicroMain 45 – Specs:
Design: 3-way active
Woofer: 8-inch aluminium cone
Mid: 2 x 2.5-inch aluminium cones
Tweeter: 1-inch ring radiator
Frequency response: 40Hz – 45kHz
Wattage: LF 250W, MF 180W, HF 180W
Dimensions: 11 x 15.5 x 11” (279 x 394 x 279mm)
Weight: 37.5lb (17kg)
Compatible Sub? Barefoot MicroSub
Available from: KMR Audio +44 (0) 20 8445 2446
10. Genelec M040
Finnish company Genelec have a fine studio monitor pedigree, stretching back to the early 80’s when their large, wall-mounted designs were the speaker of choice for a huge number of professional recording studios worldwide. Still going strong, it’s now possible to get the Genelec name in your own studio for a fraction of the price of their hefty ancestors with the brilliant, small and mighty M040 (and even smaller M030).
Genelec have thrown an impressive number of acronyms into the design of the M040. There’s the Directivity Control Waveguide (DCW™), which ensures flatness of the overall frequency response, the Laminar Integrated Port (LIP™) moulded into the cabinet to aid faithful bass reproduction, Intelligent Signal Sensing (ISS) that automatically enters standby mode if no signal is detected for 30 minutes, and the Natural Composite Enclosure (NCE™), which refers to the environmentally-friendly makeup of the cabinet itself. Elsewhere, you’ll find simple-to-use room response compensation controls, in the form of dip switches, mounted on the rear of the cabinet, to tailor the response to any acoustic environment.
Genelec M040 – Specs:
Design: 2-way active, ported
Woofer: 6.5-inch cone
Tweeter: 1-inch metal dome
Frequency response: 44Hz – 21kHz
Wattage: MF/LF 80W, class D, HF 50W, class D
Dimensions: 13.3 x 9.3 x 9″ (337 x 235 x 229mm)
Weight: 15.4lb (7.4kg)
Compatible Sub? Genelec 7040A
6 July 2018 (article by Rob Stewart)
How to create wider sounding mixes
- Stereo width is an illusion that we either capture in a recording or create within a mix
- To make wide sounding mixes, capture width when you record or create width inside your mix
What is stereo width?
Stereo width is an illusion of the left-to-right dimensions of the sound field (i.e. sound stage or panorama) in a recording, perceived by a listener.
Imagine yourself standing on a sidewalk listening to busy downtown traffic. You are hearing cars constantly rushing back and forth, left to right and right to left. This “field of sound” you are hearing is quite wide. Because we hear binaurally (another future topic), you’re hearing a three-dimensional world of sounds from left to right, front to back, and even up and down. But let’s simplify for a moment and focus on the fact that cars are coming from one direction or the other and we largely are hearing that back and forth movement across the sound field because we have two ears and we can sense where each of the sounds are coming from.
Imagine that you capture a few minutes of what you are hearing on the sidewalk with a stereo audio recorder, go back to a quiet place to listen to what you recorded. Setting aside mic quality, technique and other factors for the moment, you will hear a rough representation of what you experienced on the sidewalk because you will have captured the scene in stereo. Your recording will reveal how the traffic moves from left to right and right to left across the sound field captured by the recorder. Show that recording to someone else, who wasn’t there when you recorded it, and they will be able to imagine to a degree what it was like standing on the sidewalk because the recording has enough information in it to recreate a panorama where they can hear the traffic moving back and forth, similar to how you experienced it.
Now imagine if you use a mono (i.e. monophonic or monaural) voice recorder to capture what you heard on the sidewalk, instead. You will hear a much different representation, because your recording is missing a massive amount of information. A mono voice recorder only has one microphone (i.e. one ear) so it cannot capture stereo information. Your recording will reveal a mush of engine noise, maybe the odd vehicle horn, but everything is crammed together and it sounds like varying levels of noise. Show that recording to someone else, who wasn’t there when you recorded it, and they will find it challenging to imagine being on that sidewalk. The resulting panorama from the mono recording simply does not contain enough information to do that.
Stereo width therefore depends upon stereo information being captured and presented back to the listener. When you think about that traffic scene, what makes the scene stereo? It is all about differences between what you hear from one side of the scene to the other. Stereo width is that simple. You create stereo width by creating or enhancing the difference between the left and the right sides of the sound field that you are presenting back to your listeners.
One more example before we move on. Picture yourself sitting in front of a stage with two musicians standing side by side, a few feet apart. Forgetting the room acoustics and other factors for a moment, you focus on the two musicians, and from your vantage point, they are relatively close together. You therefore perceive them both coming largely from the center of the sound field. Now imagine that each musician walks to the opposite end of the stage. Now you perceive them as being distinctly separate where you hear one largely off to the left, and the other way off to the right. The musicians have just “widened the panorama” presented to you simply by separating themselves further, relative to you. Now imagine replacing yourself with a stereo array of microphones. See how microphone placement relative to the subject(s) can make a big difference in this situation? You can drastically alter what you present to a listener – which impacts their perception of width – simply by changing the placement of the subject(s) relative to the microphones.
This is how stereo width within a mix works. When you create a mix, you are building a sound field with layers of sound. Stereo width is an illusion of the left-to-right dimensions of the sound field. The more you can create differences between what you present on the left versus what you present on the right, the wider your mix will seem to a listener.
How to Create wide Stereo mixes
- You can only enhance stereo width if it exists to some degree in your mix
- Make your mix sound wider using differences in left vs right gain, time, pitch, tone or polarity
Mono sound is single-channel sound, and stereo (stereophonic) uses two channels. If you reproduce monophonic sound over two loudspeakers, you will hear the sound coming from a narrow area between the two speakers (in the center). This is called a phantom image because it can be quite palpable to a listener. It can sound almost as real and distinct as a sound that comes from just the left or the right speaker on its own. The phantom image sounds focused (narrow) because each loudspeaker is presenting the same information, and there is nothing that our brain can use to establish a difference between sounds coming from the left versus the right loudspeaker.
Does your mix sound too narrow? Width is a key component of modern mixes because most music is mixed in stereo. Many listeners listen in stereo (e.g. with earbuds). We perceive a mix as being too “narrow” when we sense that the music is coming from the center when it seems that it should not. That is to say, a center-centric mix can be perfectly fine if it sounds natural, but you know you have a problem when you or your listeners sense that things are too narrow, where the mix sounds too confined and the panorama is not expansive or convincing enough to draw them in.
You can make your mixes sound wider by maximizing the difference between what you present in the left versus the right channel. The “difference” can be in time (arrangement of notes or elements, and includes phase as well), gain, pitch (arrangement of notes and/or tuning), tone, or polarity.
Let’s look at gain first. Panning is used to adjust the “gain difference” between left and right. Pan hard left, and you have full gain in the left, and no gain in the right. This is the easiest way to maximize left-to-right difference. Take two different sounds in your mix, pan one hard left, the other hard right, and you have maximum separation, maximum difference between left and right. The more you “pull them in” by panning closer to center, the more you reduce difference information, bringing them closer and closer to perceived center.
You can use timing to create left-to-right difference by delaying one channel relative to the other. You can take a mono string section track in the left channel, add a 25ms delayed version to the right channel, and with some gain adjustments you have created a very wide, very stereo sounding string track. If you experiment with this, a 15ms delay is a good starting point. As you pull it towards 0ms, you will hear the delay getting harder to discern and other things start to happen as the sounds fuse together, which can be good or bad. I recommend listening in stereo for the effect you want but then listen in mono to make sure you haven’t created a mono-compatibility issue in the process. If you hear the tone change drastically where it is clear in stereo but muffled in mono, that is an issue. Also listen to the attack portions of notes. Sharper attacks (guitars, vocals etc.) will only tolerate shorter delays because if the delay is too long you’ll get a flam effect which can be distracting to a listener. Sounds with softer or slower attacks will often work well with over 20ms of delay.
You can create tonal differences between left and right by equalizing each channel differently. A common technique with mono piano tracks is to EQ the left to have more bass and lower mids and the right to have more upper mids and treble so that there is a sense of movement from left to right as the player moves up and down the scale.
You can use tuning differences between left and right to create width. This is partly how chorus effects work. Take a mono track in the left, and then add a detuned version in the right channel to create width.
A mix engineer uses all of these methods different ways to either create or enhance stereo width. The ideas are almost endless. You can keep it simple or get really creative depending upon the situation. For example, putting reverb on one side of the field but not the other is often used on guitar tracks. Another option is putting vibrato on one channel and not the other, or mono chorus on one channel or the other. Hammond’s popular “Leslie” effect, captured in stereo, works effectively because it uses the Doppler effect which essentially creates timing and tuning changes between what’s captured in the left and right microphone.
MAKE WIDER MIXES BY MAXIMIZING CONTRAST BETWEEN MONO AND STEREO
This is absolutely key. To perceive width, we need a reference point. Making every sound in your mix super-wide will not necessarily lead to an engaging or musical mix. Certain anchor points in a mix that are kept central, such as a lead vocal, a bass guitar, a kick drum etc. are essential to creating a wide mix because we will judge the width of the panorama relative to those central elements! Never underestimate the power that mono, central elements have in enhancing the perceived width of your mix by how they create contrast.
WIDENING YOUR MIX WITH SPATIAL ENHANCERS AND STEREO WIDENERS
Remember: Width must exist within your mix before you can enhance it in a meaningful way.
Please note that I consider these next tips to be shortcuts. If your goal is to truly understand stereo width to create musically wide-sounding mixes, these shortcuts will mainly help you by allowing you to easily experiment. They are a time saver that can be useful in some mix situations after you have learned the concepts above.
Stereo wideners, ambience retrieval, or other forms of spatial enhancers have become very popular. They employ different combinations of the techniques above to broaden and deepen the sound stage. They are a quick fix. I caution against using them across a whole mix because that can greatly distort the sound field you have spent so much effort to build if you are not careful. I have lumped ambience retrieval in with stereo widening but they are often different processes, so research any processor first to understand what it will do to the sound. Use these products sparingly, and consider only using them on certain elements within your mix that would benefit from widening or ambience retrieval effects.
“FreeHaas” and “FreeOutsider” are free plugins offered by VescoFX. They are well worth experimenting with. I recommend using them sparingly, on one or two elements in your mix at the most. FreeHaas adds a Haas Delay (see Haas effect) which you can adjust to your liking. FreeOutsider is a much more obvious beyond-the-speakers effect. You can combine them with tone and gain adjustments to maximize the difference between left and right, thereby maximizing the width of certain elements of your mix.
Mid/Side (a.k.a. sum and difference) processors are widely misunderstood and are often thought of for stereo widening, but I caution that merely adjusting the ratio of Mid to Side will not work that well unless the source has a lot of difference information in it already. This is why many spatial enhancers include a Mid/Side adjustment control to allow you to adjust how much of the widening effect you hear, after the spatial processor has done the initial work of creating more difference information in the first place. If your mix has a lot of difference information, but does not sound wide enough to you, Mid/Side adjustments probably cannot help. If your source is completely mono, Mid/Side will do absolutely nothing and if it is close to mono, then boosting the side doesn’t do much either. I expand upon this much further in Part 1 and Part 2 of my Mid/Side articles.
Widening your mix beyond the loudspeakers
While much of the effort in a mix is in creating a wide field within the loudspeakers, you can create the illusion of going past them as well, to further extremes. There is risk to this. If you overdo it, it will lower the quality of your mix.
Take any sound and add it to two channels in your workstation. Pan one channel hard left, pan the other hard right. Press play and you’ll hear the sound coming from the exact center. Flip the polarity of one of the channels and listen to how the sound changes. It will sound strange – very wide depending upon your listening environment. The reason is because yet again, you have created another difference between left and right – this time it is a difference in polarity! This is a completely unnatural sound and for most people it is fatiguing after a while. But it is also completely incompatible in mono. Collapse your mix bus to mono, and everything disappears because the left and right channels cancel each other out. While absolute polarity differences like this are unnatural, they can also be useful. Experiment with them, particularly on occasional ambient sounds or even reverb effects, and with the other techniques that I mentioned above to help push the sounds a little further outside the speakers.
By combining various techniques, we can recreate more complex combinations of binaural cues that our brain uses to determine where a sound is located, allowing you to present a three-dimensional sound stage to your listeners. This is worthy of a separate article but here is a simple example. When we hear a sound coming from our left, our left ear hears it slightly louder, slightly sooner, and slightly brighter than our right ear does. Our brain uses that combination of differences in loudness, timing and tone to perceive where that sound is located within a three-dimensional sound field. Our brain analyzes that, along with the information about ambient environment (early reflections and the left-vs-right loudness, timing and tonal differences of sounds in the environment) to map out how large a space is and where the sound is located within it. The more location information we can present to our brain, the more we can create the perception of location be it within or beyond the loudspeakers.
It is not easy to recreate these types of cues over loudspeakers because of bleed (crosstalk). If you sit between any two loudspeakers, you will hear both loudspeakers with both ears to varying degrees whereas with headphones, each ear only hears one speaker. It is therefore much easier to recreate these complex cues when you are mixing for playback over headphones because you won’t have to overcome crosstalk. Creating these cues for headphones can be as easy as using a binaural microphone array to capture the source, and then working to preserve those cues throughout the production process. You can also add the cues after the fact with an HRTF (Head Related Transfer Function) processor.
When mixing for playback over loudspeakers, the only way to effectively recreate these cues is to find ways to minimize the perceived crosstalk. There are crosstalk cancellation technologies available such as QSound, Ambiophonics and BACCH that accomplish this to various degrees. I will plan expand more on this in a future article. These types of filters are well suited to situations where your listeners are sitting in a fixed position, in front of two loudspeakers which makes them well suited to gaming applications or small portable devices. BACCH might be the most promising from a pure performance perspective but unfortunately at the time of this writing it is priced outside of the range of most users and productions.
Outside of crosstalk, there are other challenges to reproducing a three-dimensional sound field over two loudspeakers. There is no way to know exactly where your listener will be placed, but we can be certain that most will not stay in the same position. We also cannot know what performance level their listening room and sound system is capable of, whether it is mono, stereo or surround. We can’t even guess at how it will be configured (EQ or tone, speaker placement etc.). Therefore, if you plan to build complex binaural cues into your mixes, carefully consider all of the possible end points (ear buds, loudspeakers, television, movie theater etc.), and how your music will be consumed (while gaming, while driving, while travelling, while housecleaning). In many cases, building these complex positional cues into your mix is only worth the effort if you can do so without reducing the sound quality for other listeners in the process. This is why checking for mono compatibility and loudspeaker vs headphone compatibility is so important.
24 June 2018 (article by MusicRadar.com)
Inspired by the likes of Philip Glass and Brian Eno, ambient music is as much about creating mood as it is creating melody.
Fortunately, computer users can now call upon an arsenal of ambient-friendly production tools – MusicRadar is here to explain what they are and how to use them.
1. If all the soft sounds and smooth vibes get a little too much, try some juxtaposition. Ambient heroes The Orb are fond of this technique, and whether it’s a squealing guitar, devastating synth hit or ridiculous vocal sample, they’re not afraid to toss something a little unusual into the mix.
2. Getting off-the-wall sounds doesn’t have to involve spending hundreds on sample downloads and libraries – there are plenty of interesting sounds happening all around us all the time. If you’ve got a mic and a laptop – or any portable recorder – take a field trip and record some of nature’s bounty. Running water’s always good for a laugh, but remember: your equipment should stay dry, even if you don’t…
3. Second-hand record shops are great places to find sounds. You may even find that your local charity shop has an untapped collection of oddities just waiting to be snapped up by the enterprising samplist. From records featuring nothing but steam engine noises to children’s story albums, there’s an abundance of weirdness out there for the taking.
4. Samples are a constant source of inspiration, but it’s easy to discount one because it doesn’t fit the feel of your track when you first try it. If you’re short on fresh ideas, try running short bursts of a sample through a delay effect. Using this method, it’s possible to come up with some great abstract noises that sound nothing like the original source material.
5. If your tracks are jam-packed full of synthetic-sounding virtual instrument patches and everything’s starting to sound too ‘computery’, consider bringing in some natural sounds or using a few real instrument parts. Even if they’re from ROMplers, it should help take some of the unnatural edge off.
6. Recordings of natural sounds such as rainfall, waves, wind and fire are great for filling out a mix because they’re basically noise, and as such, they have a wide range of frequencies. They shouldn’t be too loud or they’ll overpower the mix, but use them with care and they can be extremely useful.
7. Noise is a useful synthesis tool – if your synth features a noise oscillator, you can use it with a fast-attack amplitude envelope to create your own percussion sounds. This sounds artificial, but in a lo-fi way, and works especially well when teamed with a high-quality reverb.
8. If you’re using long, sustained sounds, such as pads, your mix can lack movement if these elements are too static. By subtly altering tuning, pulse width or filter cutoff over time, you can create more organic sounds that will enhance the mix rather than make it sound lifeless.
“Recordings of natural sounds such as rainfall, waves, wind and fire are great for filling out a mix because they’re basically noise”
9. If you’ve got a sample that you want to play for longer than its duration, you have two basic options: you could timestretch it, which will most likely introduce unwanted audio artifacts, or loop it. Crossfade looping is the best way to get seamless loops, but if this isn’t possible, you can recreate the effect yourself by fading between two audio tracks in your mixer.
10. To make a pad sound particularly evocative, try modulating the filter cutoff with a shallow LFO as well as a big, sweeping envelope. This will give the sound a great deal of movement and works superbly when combined with a delay effect.
11. When working with vocals, you can have a lot of fun with pitchshifting. When pitching vocals around, it helps to use a plug-in with a formant control – this helps vocals retain their characteristics or, conversely, can be used to alter them radically. Check out Smoky Joe, a lo-fi formant processor.
12. With modern audio sequencers, it’s easier than ever to cut up vocals and other rhythmic sounds in order to fit them in with the groove of your track. When cutting sounds up in your sequencer, remember to zoom in to make sure you’re cutting the file at a point where the amplitude is zero – otherwise known as a ‘zero crossing’.
13. When deploying your newly-sliced rhythmic samples, it’s not always best to have your sequencer’s snap control active. You might find that pulling samples forwards along the track a little makes them fit in better with the rest of the groove, and having the snap control turned off also makes programming human-sounding rhythms easier.
14. Silky bass guitar tones are a common sound in ambient dub, but if you don’t have a real bass guitar to hand, you’ll have some trouble getting the same smooth sound. Bass ROMplers such as Spectrasonics Trilogy and Bornemark’s Broomstick Bass are your best bets for recreating this kind of thing.
15. Whether you’re composing in stereo or surround, it’s important to use the available panoramic space properly if you want to create a sense of size. If your track has drums, you’ll probably want to pan these around the centre, but with synths and effects you can afford to use the space more creatively, so try panning them around.
16. Most DAWs have simple pan controls that only enable you to pick one position in the stereo panorama. If you’re looking for slightly more control, a stereo imaging plug-in such as mda Image or BetaBugs Moneo can be used to control the position and filter setting of each channel or tweak them as a mid/side pair, respectively.
17. To add a natural stereo panorama to mono samples, you could do a lot worse than give Voxengo Stereo Touch a try. This effect uses a delay algorithm to create a convincing stereo effect that’s guaranteed to revitalise any dodgy old mono sounds you might have lying around.
18. Reverb is one of the most important tools you have for creating a sense of space, so if you’re making ambient music, it pays to take the time to get it as sweet as you can. A good start is to use a high quality reverb – Ambience isn’t just free, it’s one of the best reverb plug-ins out there.
19. It can be tempting to just stick reverb on a few tracks and leave it at that, but that wouldn’t be using this powerful effect to its full potential. Using high damping values, large room sizes and long reverb times will create a big sound that, when combined with judicious EQ, can create a ‘far away’ kind of effect.
20. When using reverbs, if you want to create a softer, more ethereal effect, use less of the dry signal in the output. You can do this by turning the wet/dry ratio up, or, if you’re using a send effect, by setting it to pre-fader and turning the source channel’s main volume level down.
21. If you’d rather have a brighter, closer effect, then make the reverb’s damping less severe, reduce the room size and turn down the delay time. This works especially well in conjunction with stereo enhancer effects such as the Voxengo Stereo Touch plug-in.
22. Many interesting effects can be created by rendering out reverb and delay tails minus the original dry sound, then applying creative processing to the tail. Filters work particularly well for this kind of thing and, once processed, the new sound can be played back alongside the original version, or replace it altogether.
23. Finally, when programming synth patches, don’t discount the creative potential of your instrument’s reverb section. With a long, lush reverb, even the smallest synth squelches or blips can be turned into pleasingly tonal atmospheric effects. Of course, if your synth effects truly suck, you can always use a separate reverb or delay plug-in instead to create the same effects.
24. Delay is a pretty common effect in atmospheric music like ambient, but for ambient dub, a full-on feedback delay, such as Ohm Force’s excellent OhmBoyz effect, is just the thing.
25. Dynamic use of feedback delay is useful for creating long, evolving rhythmic effects. By automating the feedback control on a delay plug-in, you can build to a crescendo or create weird rhythmic effects.
26. Getting that distinctive morphing dub delay effect can be done by adding either a filter or distortion component to the feedback loop – easily done in OhmBoyz, as it has both. If you’re using a delay effect in Reaktor or another modular environment, you can add these elements yourself, though it’s advisable to put a level limiter after them to ensure the feedback doesn’t get out of control.
27. Delay effects work well before a reverb, though too much of either will swamp the mix. However, it’s possible to tame these effects with automation – set the reverb’s wet level to 0%, automating it so that it comes up as the end of the delay tail is playing. This way, you’ll be able to use both the delay and the reverb, without having too much of either going on at once. As an advanced alternative, you could use sidechain compression to duck the start of the reverb (using the source signal as the key input), and setting the release time appropriately, thus achieving the same effect automatically.
28 May 2018
How to Manage & Process Sub-Bass Frequencies
by Mo Volans
Whether you are looking to add or remove sub-bass frequencies from your mix, it’s essential you’re armed with the right tools and techniques. Everything from a simple EQ to multi-band treatments and enhancers can be used, so there are certainly a few routes to choose from here.
If you are in the business of creating club music of any kind, nailing these all important lows is a must. Get it right and your track will rock, leave them unchecked and your mix is in danger of becoming colored and your master distorted and quiet.
Step 1 – Monitoring Sub-Bass
If you’re planning on making bass-heavy music with frequencies that register anywhere below about 50 – 60hz, the first thing you should do is check that your monitors are actually able to reproduce sounds in this range. There is no point in using processors to enhance or cut in this area if you can’t here what you are doing.
Some larger monitors will go as low as 35-40hz giving you a pretty clear representation of what is happening with your subsonics but smaller monitors might only be capable of producing lows down to about 50 or even 60hz. This really leaves a huge gap in your mix. These frequencies may not be critical in certain genres of music but when it comes to music destined for the dance floor, they are nothing short of essential.
Of course some of you may work in an environment where small monitors are a must to keep down noise levels, or budget maybe an issue so larger speakers may not be a viable option. If this is the case it may be worth thinking about mixing your music elsewhere, maybe a friend has a studio with a larger set of monitors where you can perform your final mix-down. Unfortunately the bottom line is these frequencies have to be heard if your mix is to work well on a large system.
Another solution is to acquire a pair of monitors that are capable of reaching the low frequencies needed here. There are a few issues though; in order to reproduce sub-bass, speakers tend to utilize large bass drivers. 8 or 10 inch drivers are often required for the job and this tends to make the cabinets pretty large, which can be an issue if desktop space is limited. Large speakers can also present problems when used in smaller spaces and in close proximity to the listener.
Arguably the best solution to evade all these issues is to install a dedicated sub woofer, and although this does involve some expense it is certainly the most streamlined method for reproducing very low frequencies. Professional quality subs can be extremely expensive but there are products on the market now that deliver a lot of bang for your buck and sport features that may not have been available even a few years ago. Most modern sub woofers will play back frequencies as low as 20-30hz with ease.
If you do decide to use a dedicated sub then there are a few things you should look out for. An active sub is most likely to be the sensible choice for the modern digital set up as the amp is not only integrated into the unit but also matched to the drivers in the cabinet, a protection circuit is often present in this configuration, which helps avoid blowing anything up at high volume.
Other features to look out for are internal crossovers which allow you to connect your main speakers to the sub and handy extras such as bypass foot switches for taking the sub out of the mix. Companies such as M-Audio and KRK are producing products with all these features for very reasonable prices.
Step 2 – Using EQ to Control Low Frequencies
Once you have decided on the best way to monitor your sub-bass frequencies you will be able to hear exactly what you are doing across the whole frequency range. Assuming that you are good to go in this area, let’s have a look at some different tools and techniques we can use to manipulate our low end.
Perhaps the most straight forward way to control the subs in a sound is to use a simple shelving EQ. This may seem a little obvious but once you have a monitor set up that produces low frequencies that were previously missing you may be surprised the effect a small amount of EQ has on a sound. As most EQ plug-ins go all the way down to 20hz it is pretty simple to add a few db of boost or attenuation to a sound in the area of 20-80hz.
When you first start to focus on these lower frequencies it can be a good idea to use a spectrum analyzer to get some visual feedback on the changes you are making using the EQ. This, combined with some critical listening, can really help to fine-tune your ear to the subtle differences you will experience here. Some EQ plug-ins such as Logic Pro’s actually include a spectrum analyzer within its interface.
As with most forms of processing the majority of us strive for transparency and attempt to avoid over-coloring our sound. With this in mind, using a shelving EQ to mildly boost or cut the subs in our mix can be preferable to using an enhancer or exciter for the job. Sometimes of course these more intense methods are necessary but it’s worth trying a more subtle approach first.
Logic EQ and Analyzer
UAD Cambridge EQ
Step 3 – Using Filters to Remove Subs
Some sounds in our mixes will need to have their very low frequencies removed, to let other elements shine in this area. For instance a sampled guitar loop may need to have some low end removed to allow a kick drum or bass part to really stand out. You should really think about each sound in relation to the rest of your mix and ask yourself if its low frequencies should be removed or left intact. Closely managing your sub frequencies in this way will help you create a clear and focused low end mix, with real power.
Even sounds that have had their low frequencies boosted can sometimes benefit from having some sub-bass removed. Most sound systems, even very large ones, don’t tend to produce frequencies much below 30hz, so having a lot of energy present in this area is not really necessary. This means you could be boosting 30-80hz and cutting below 30hz at the same time. This sort of treatment can also be useful in mastering.
When it comes to the right tool for cutting these lows it should be pretty obvious that a standard shelving EQ just isn’t powerful enough, so we have to turn to high pass filtering for the task. A filter is a lot more abrupt than your average EQ and deals in absolute values. If we set a high pass filter to 200Hz, all signal below 200Hz will be removed. Filters can be pretty extreme but in some cases this is exactly what is needed. Saying this, many filter plug-ins will allow the curve to be altered thus reducing the intensity of the effect.
When looking for the right filter plug-in to use, there are generally a couple of types to choose from. You will find many dedicated filters are resonant models, which work in pretty much the same way as the filter section on a synth. These are fine for removing specific frequencies from a sound as long as they are used without any resonance. Fabfilter produce an excellent filter plug-in called ‘Simplon’ which is perfect for removing low frequencies and features a few different curves.
If you find these resonant filters are too harsh or color your sound in any way, you can opt for something a little more musical. Many EQs, in both hardware and software form, feature built in high pass filters and these are often a lot more subtle and not quite as severe as their resonant counterparts. Try this approach when working with more organic sounds, such as recordings of acoustic instruments and vocals.
Neve 88RS Filters
Step 4 – Multi-band Processing
An alternative to using EQ and filters is multi-band processing, specifically multi-band compression. There are a huge number of excellent multi-band compressors on the market now and many DAWs such as Cubase and Logic include them as bundled stock plug-ins.
The good thing about these processors is not just the transparent sound they deliver but also the fact that they are capable of adding or subtracting sub-bass frequencies.
By simply adjusting the lower band of your multi-band compressor to cover everything from 20hz – 80hz you are able to process the sub frequencies in total isolation. Because these plug-ins are dynamics processors you also have the ability to tweak ratio, threshold, attack and release of the effect, as you would have in a standard compressor. This extra control creates the perfect environment for controlling sub-bass.
Step 5 – Sub-Bass Enhancers
The final type of processor we will look at here is the dedicated sub-bass enhancer. The reason I have left this until the end is that in my opinion it should really be used as a last resort. Not that this family of processors don’t often have a lot to offer but as with any other enhancer there is a danger that if you are not 100% sure what is happening under the hood, you could be doing more damage than good.
Of course it is likely that you get what you pay for here and products from companies such as MaxxBass from Waves or even Logic Pro’s Sub-Bass can produce excellent results but some free plug-ins offering similar algorithms may play havoc with your low end mix. This doesn’t mean there aren’t some excellent free plug-ins out there, it just may be wise to choose very carefully in this area.
If you feel that all other methods just aren’t cutting it, then you may want to try one of these plug-ins and if you find some of them a little intimidating or just fancy, try something with very few controls. Look at the classic BBE Sonic Maximizer, now revamped and market by Nomad plug-ins. With only two knobs this little processor can work wonders on both the low and high end of any audio fed through it.
BBE Sonic Maximizer
Step 6 – Layering Sounds
If you want to try a technique that involves little to no processing to achieve a boost in your sub-bass frequencies, you could experiment with layering different parts. This can work with everything from bass lines to kick drums and percussion. It’s simply a case of adding a duplicate MIDI track under the original part and pointing it at a new instrument that is generating a low frequency rich sound.
For example you may feel that a bass part doesn’t pack enough low end punch, simply duplicate the MIDI and use it to control a synth with a sine wave bass patch pitched one octave lower. Mixed carefully this simple technique can add all the bass you need without any processing at all.
If you find yourself wanting to use a similar method but your bass line is an audio loop or sample, you may still be able to achieve the same result. Many DAWs now include technology that allow you to convert audio to MIDI. Simply put, this should allow you to generate a MIDI file from any pitched audio sequence. Of course, the quality of results will vary from one case to another but it is certainly an option here.
21, April 2018
REVERBS FOR MULTI DIMENSIONAL SOUND
Reverb is a classic mixing tool for adding width, but also that third dimension to your mix: depth.
By adding depth to your stereo image, you’re also expanding the stereo image as a whole. Reverb will give you more room for every sound to breathe and settle into the mix.
There are many different ways to use reverb and add space to your mix, but any reverb technique will add some degree of depth and spaciousness to your mix. And there are many types of reverb. Each is capable of adding a distinct vibe and depth to your mix.
Choosing the perfect type of reverb to give that extra space without drastically changing your audio’s character will take some practice. But when it comes to width, Hall reverb is a good place to start.
Don’t stop there though… all types of reverb can do wonders for adding three-dimensionality depending on your mix and production style. It can be useful to experiment with different reverbs for different tracks in the mix, or alternate dry tracks with reverb treated tracks. With small amount of effect, that can add unpredictable and variable spaciousness during the final mix.
Hot Tip: Using reverb with a short decay time will add a subtler reverb effect. It’s great for when you want to add width and depth without changing the overall character of a sound.
Some examples where reverbs are used with a creative and functional approach:
14, April 2018
Dark Ambient technicalities. Michael Barnett interviews Sonologyst about creating dark ambient.
(extracted from the Michael Barnett article: Dark Ambient 101: Understanding the Technicalities – http://www.thisisdarkness.com/2018/03/17/dark-ambient-101/ ).
- Analog or Digital?
The mix of both of them is ideal.
M.B.: What do you see as the differences between analog and digital creations of dark ambient music?
S.: There’s no difference from a creative point of view. Obviously there are a lot from the technical one.
M.B.: What are some of the key instruments/programs that you use to make analog dark ambient?
S.: Analog synthesizers, electrified string instruments, guitars, samples, editing software and plug ins, percussions, wind instruments parts (commissioned to other musicians), tapes, pedals, dronin.
M.B.: What are some of the key instruments/programs that you use to make digital dark ambient?
S.: Mainly plugins to work on noise parts and editing softwares.
M.B.: Do you see one or the other as being the “better” technique for creation of dark ambient music?
S.: Everyone has to develop the better process fitting with her/him attitude.
- Drones? M.B.: What are some of the techniques you use to create drones?
S.: There are different ones, maybe infinite. It’s possible to make drones with stratifications of synth pads, by editing acoustic instruments like brasses, winds, string instruments and so on; playing heavily distorted bass and/or guitar; editing samples, using noise from modular synthesizers, editing field recording, recording the washing machine noise and on and on..
M.B.: Do you have a favorite program/instrument to use for creating drones?
S.: Not a specific one.
M.B.: As a beginner did you create drones the same way you do now?
S.: As a beginner I made a lot of mistakes before to find my way.
M.B.: Have you changed techniques/software/instruments for creating drones over the progress of your career?
S.: Yes I did it many times. And I continue to change to make the sound fabric different in any production I do.
M.B.: How important are drones to dark ambient music?
Probably drones are the dark ambient trade mark, as well the violin and piano are in the classical music for orchestra, or the electric guitar solos are in the rock music.
- Field Recordings?
M.B.: How important are field recordings to dark ambient music?
S.: They are another fundamental component in dark ambient music. They are the ingredient to create visual atmospheres, vivid landscapes, even stories, and forge a solid concept when the musician has something interesting to tell through the music.
M.B.: What electronics do you use to capture field recordings?
S.: I’m not a professional of field recordings, so I use simply an IPhone when I’m around to catch everything could be interesting.
M.B.: Do you leave the field recordings raw or do you add effects treatment to them?
S.: I usually treat field recordings with additional reverbs. But the most important thing is to find the right level for the field recording layer in the mix. Mixing is by all means a crucial part in the process.
M.B.: Do you use field recordings in the creation of drone or do you only use them as a secondary layer of sound?
S.: It’s a possible choice to use f.r. for drones, why not?
M.B.: Do you use human vocals in dark ambient?
S.: Yes human vocals.
M.B.: How important are human vocals to dark ambient?
S.: It depends of the concept behind the work, but I find human vocals important in my music, especially the spoken words.
M.B.: Do you create your own passages to recite?
S.: Just sometimes.
M.B.: Do you use your own voice, hire a voice actor, or use samples from films/television/speeches?
S.: Yes, samples from old documentaries, movies, speeches are my favorite. But I also asked singers to send to me parts for specific uses.
M.B.: Is it necessary to ask permission of the original copyright holder before using samples of vocals in your music?
S.: No, for they are usually very short samples or free samples.
I prefer to escape all questions about DAW, computer and so on, simply because there are not peculiarities for dark ambient music. The logic of hardware and software is the same for all kind of music. Just I can add that I’m a graduated sound technician, so I learned technique of recording, mixing and mastering through regular courses. But as in all studies, the experience is the most important factor. Do it, do it and do it again. And after some years everyone will find the right set up and process. And for people like me, who don’t have big amounts of money to invest in expensive hardware and software, the experience will help to do more, using less. And this is a big advantage for creativity; when you have poor instruments and have to use your brain to find out something good. Take a few small stones, beat each other and record the sound by using some freeware delay and reverb. Probably you will be very positively surprised of the result.
M.B.: Where do you go to find samples?
S.: Everywhere: everyday life, music, movies, documentaries, vinyl, VHS, NASA web site, specialized platforms for samples sharing…
M.B.: What samples would be off-limits in a legal sense?
S.: There are a lot of free samples around, or simply usable by asking the owner permission. But for more specific knowledge of the argument I suggest to read the related laws of the source origin country.
M.B.: How do you extract samples from movies, games, speechs?
S.: Through Youtube when it’s possible by a software, but for more original sources by connecting the source (turntable, VHS player, microphones,…) to the audio interface.
M.B.: How important are samples to dark ambient music?
S.: Important, but not necessary.
- Instruments:M.B.: When you use instruments in your music do you play a real instrument yourself?S.: Yes I do it. M.B.: If you want to have violin, for exampe, (or any other instrument) in a song, but don’t own one and can’t play one, is there another option? (some sort of program that will create violin sounds for you?)
S.: I prefer to directly ask other musicians to realize the part, so to have a more natural and warm sound effect.
- Mastering M.B.: How important is mastering in dark ambient?S.: It’s fundamental.
M.B.: Can a musician master their own album with limited training?
S.: It’s not an easy job without a little of training.
M.B.:What programs do you use to master an album?
S.: I prefer to not tell that. It risks to be a commercial advertising for software companies
M.B.:If paying another person to master an album, what credentials should they have? (ie. do they need to make dark ambient themselves to understand how to master dark ambient?)
S.: It would be better if the mastering service comes from a person with a good sensibility for that kind of music. If a musician who plays himself that music, that’s even better.
M.B.: What are the differences between mastering an album that is digital, CD, cassette or vinyl? Should each have a separate mastering?
S.: There’s a certain difference about mastering a vinyl compared with cd or cassette mastering. It’s related to the output levels that differ, in the vinyl case, depending of the track position (closer to the edge or the center). So as matter of fact, they are two completely different mastering. But for this I suggest to read this article from Sonologyst blog: https://wordpress.com/post/sonologyst.com/168
- General AdviceM.B.: What advice would you give to a person just coming into dark ambient as a potential artist?S.: Just to work with passion and not to be hurry, releasing huge amount of music, just to show the audience what is going on. That is a mistake many people do, while the process to improve the own style should be something private. M.B.: What are the best aspects of creating dark ambient? S.: It gives to you the possibility to be in deep connection with you profound states of mind. M.B.: What are the worst/hardest aspects of creating dark ambient?
S.: There aren’t worst aspect to me. M.B.: What are somethings an amatuer should avoid doing at all costs?
S.: I replied to this question in the previous point. M.B.: How frequently should an artist aim for releasing albums (several times a year?, once a year?, once a month?)
S.: Every artist has to find the own way for that. It’s impossible to give a general advice. In my case I found the good and natural rhythm working on one release a year. And I don’t exclude to increase the interval between two works. That lets me a major deepness, awareness and consciousness of what I’m going to do. Basically I start a work when I have really something to communicate, and after I’m aware of that, I need time to explore how to communicate it.
M.B.: Should a musician know the history of the genre before creating their own music?
S.: Not necessarily, but it would be a crime to ignore all that beautiful music created in latest decades.
06, April 2018
10 Tips for reverb using by John Griffin
Reverb is necessary in order to create the impression of distance and separation between elements, but it also contributes a lot to the ‘glamour factor’ you’ll need for a modern commercial production. Quite simply, making the wrong reverb choices is a strong indicator of a non-professional mix. It is potentially so destructive that many of us are either too conservative when we use it – resulting in no real benefit – or else it’s applied too liberally and smears over all your previous delicate mixing manoeuvres.
What follows then are 10 essential tips to help you steer clear of the pitfalls and build a more effective reverb workflow:
1. Long and Short Reverbs
A good general piece of advice would be to use short reverbs in busy mixes, longer reverbs in music with more space. What can be deceiving though is to judge the validity of reverbs by name i.e Halls and Chambers as long, plates and rooms as short – if you think you need short reverbs you could find exactly what you want from a short Hall and you could find just the tail your after in a mix using a long plate. Rather than thinking purely in terms of long and short, think in terms of the quality of the tails, longer tails can disguise the presence of reverb where short ones can draw attention to it.
2. Pre Delay
Pre-delay is the single most powerful feature in most reverbs, setting a pre-delay allows for a certain amount of dry signal to get through before it is washed in reverb, this means greater intelligibility. It is particularly useful for keeping the attack of words from a lead vocal upfront and clear. In this sense it is much like how you use a compressor, setting a slower attack time or in the case of reverb – pre delay, lets vocal information through which is re-assuring to listen to. Anything from 20ms to about 80ms will be the area you need to work in – beyond this you will create a distinctive slap back effect that could be cool in the right circumstances but less suitable for most.
3. 3D Reverb
For a far more dimensional reverb and richer spaces you should try using multiple reverbs together. For example, using three you can create a much more convincing ambience.
To get you started – first find a small room reverb, this to give a little air around the source, second use a plate reverb and blend it as you might pour sauce into pasta, adding flavour. Third use a hall with a long tail to add ceiling, be careful not to overdo it. How do you know when you’re overdoing it? Read on…
4. How much?
It is a taste thing of course, if the reverb is a big feature of your production ala Phil Spector, you’ll use a lot more than if you were just using it to blend, glue and create a believable ambiance. Remember that a little reverb goes a long way. When it comes to applying reverb, solo the instrument or voice, then bring in the reverb until you can hear it, then back it off a smidge until you sort of feel you want a bit more. That’ll probably be about right.
5. High Pass Filtering
Hopefully you are aware of the benefits of high and low pass filters, and you’ll have been applying these to instruments in your mix already to keep low end rumble and other toxic frequencies at bay. The same wisdom works on reverb, high passing reverb with an EQ – i.e. rolling off its low end will keep the space open. Leaving the low end in, could mean you lose definition as you add more reverb to more channels.
6. Brightening with Reverb
Reverb can be especially useful for brightening vocals where you might feel that EQing the vocal directly is working against you. This same approach can of course be used on any instrument, the trick is to identify the presence EQ range and then boost that in the reverb. For example, vocals usually have a strong presence around 3K, so, rather than EQ the vocal audio, instead, insert an EQ after your reverb and push that in the 3K area to get a more airy, transparent lift.
7. Impact with Reverb
Snare drums can regularly benefit from gated reverb, whereby a half second or so pre-delay followed then by a reverb that is gated, i.e. cut short abruptly – creates an un-natural, but useful artefact that might be described as smashed glass. The effect if used proportionally and blended well will give dimension to the snare without adding body. Used heavily it is a very distinctive effect brought to popular consciousness by Phil Collins and David Bowie, however, despite how dated it can make a drum sound, it is still used a lot for sound re-enforcement even in today’s most cutting edge productions.
8. Grouping Reverb
A useful tip for gauging the effectiveness of your reverb, especially if you are using multiple reverbs is to group them together so you can then solo or mute them with a single mouse click. Being able to A/B processing in this way can be very informative and help you reign in your levels or feel confident about adding more. Having control of all your reverbs on a single fader will allow you to fine tune how you want them, plus you can of course apply EQ as mentioned in tips 5 & 6.
9. Springs and other ‘Dirty’ Verbs
Certain instruments take better to reverb than others, some don’t play so nicely. Often we like electric guitars to feel upfront, but regular reverb tends to soften their impact. Spring reverb and certain other lo-fi processors work especially well, the harshness of them can add body and presence to the audio, don’t discount cheap sounding processors and springs for use on vocals either, for a vintage lo-fi vocal reverb springs are extremely fashionable right now.
10. Mono Reverb
We often think of reverb being stereo but there is huge benefit to setting up mono reverbs. Mono reverbs are great for spot lighting or where you want to draw attention to an instrument without swamping the mix. If you wanted to spot light a keyboard solo that was panned off to the right, setting up a mono dedicated processor and then the pan to match the pan setting of the keyboard, will really help retain the dynamics in your production.
Reverb fashions come and go and presently we seem to be in a phase of rediscovering reverb. Reverb today is generally very musical and subtle: We want it, but we don’t want it to be over bearing. Modern vocal tracks and spot FX will tolerate a good deal, but we like drums to feel natural, textural and upfront. The main thing to keep in mind is that successful reverb should enhance the mood you are aiming for; it is not just about adding size, but embellishment of your central theme.
24, March 2018
MASTERING FOR VINYL
Obviously a vinyl record is a different thing from a CD or a WAV file, but does it require a separate, dedicated master, or are the two formats basically made from the same mastered file?
The answer is YES! Making a mix post production for vinyl means to make an alternate version after the digital one, with little to no digital peak limiting, and a little more headroom in the analog domain. Sending a loud and aggressive CD master to a lathe will only cause the cutting engineer to have to turn things down significantly, and in many cases they’ll be forced to cut an even quieter record than they would have with a more dynamic premaster. Your mastering for vinyl doesn’t need to be as loud as your CD master because the volume of your vinyl will be determined by the length of the sides, which means to keep louder more aggressive material near the outer edge of the record (early in the side sequence), and the more subdued and less aggressive tracks near the inner grooves (where noise and distortion become more of a consideration).
A digital master for CD has to have a 16-bit word length, and it can be as loud and as limited as the client’s taste or insecurity dictates; with the vinyl master there is a physical limit to what can be fed to the cutting head of the lathe, and so heavily clipped masters are not welcome and can only be accommodated, if at all, by serious level reduction. For vinyl, the optimum source is 24-bit. Things can get more tricky if the primary focus is the digital master, and especially when that is required to be fairly loud. You can’t simply take an unlimited file and add 4 or 6 dB of limiting without sonic consequences, and so for loud CD masters, we normally add another step of gain-staging and include some light limiting during the initial processing run, the result being a louder master to begin with for the second stage of adding gain. In this case the difference between CD/digital mastering and the vinyl one will be more relevant.
So before to go on with an alternate version for vinyl mastering, think about what’s the priority in your music distribution strategy and then talk about that with the sound technician who will care about your project.
What Is Mastering?
Mastering is the term most commonly used to refer to the process of taking an audio mix and preparing it for distribution. There are several considerations in this process: unifying the sound of a record, maintaining consistency across an album, and preparing for distribution.
The Sound of a Record
The goal of this step is to correct mix balance issues and enhance particular sonic characteristics, taking a good mix (usually in the form of a stereo file) and putting the final touches on it. This can involve adjusting levels and general “sweetening” of the mix. Think of it as the difference between a good-sounding mix and a professional-sounding, finished master.
This process can involve adding broad equalization, applying compression, limiting, etc. This is often actually referred to as “premastering” in the world of LP and CD replication, but let’s refer to it as mastering for simplicity.
Consistency Across an Album
Consideration also has to be made for how the individual tracks work together when played one after another in an album sequence. Is there a consistent sound? Are the levels matched? Does the collection have a common “character” and play back evenly so that the listener doesn’t have to adjust the volume?
This process is generally included in the previous step, with the additional evaluation of how individual tracks sound in sequence and in relation to each other. This doesn’t mean that you simply make one preset and use it on all your tracks so that they have a consistent sound. Instead, the goal is to reconcile the differences between tracks while maintaining (or even enhancing) the character of each of them, which will most likely mean different settings for different tracks.
Preparation for Distribution
The final step usually involves preparing the song or sequence of songs for download, manufacturing and/or duplication/replication. This step varies depending on the intended delivery format. In the case of a CD, it can mean converting to 16 bit/44.1 kHz audio through resampling and/or dithering, and setting track indexes, track gaps, PQ codes, and other CD-specific markings. For web-centered distribution, you might need to adjust the levels to prepare for conversion to AAC, MP3 or hi-resolution files and include the required metadata.
The History of Mastering
The earliest forms of mainstream recording technology did not require the recording, mixing, and mastering processes to be separate disciplines.
Rather, the recording was cut directly to a wax disc via a stylus connected to a diaphragm, which was in turn driven by an acoustic horn through which the sound was captured. These wax discs were then used to make stampers, which themselves were used to press shellac-composite 78 rpm discs.
The introduction of the 331/2 rpm long play (LP) vinyl record in 1948, and the 45 rpm in 1949 contributed to a change over time in the record making process. Recordings were being made to tape and engineers were tasked with preparing a master disc from the tape recording. When cutting master discs, these engineers now had to watch for and reduce loud transient peaks present in the tape recording. The energy of these peaks could potentially burn out the disc cutter head or cause the stylus to pop out of the groove when the record was playing.
In order to detect and reduce these peaks, dynamics processing tools such as compressors and limiters were introduced. This was the first time sonic adjustments began to impact the audio after the recording and mixing processes. The need to monitor these tools and adjust the settings for an optimum playback experience without compromising the sound quality was the earliest form of mastering.
The introduction of the standardized RIAA curve meant that equalization (EQ) became part of the mastering discussion. Intended to allow records to be cut with narrower, tighter grooves (and thus, a longer playing time), one side effect of this curve was that the pre-emphasis curve applied to the recording could enhance high frequency transient peaks, and the de-emphasis applied upon playback could cause a boost in low frequency energy that would cause the stylus to pop out of the groove.
Slowly but surely, the necessity of these tools to ensure a positive consumer experience meant that the skills of those who could utilize them effectively became highly prized. Some engineers (notably Doug Sax, Bob Ludwig, Bob Katz, Bernie Grundman and others) began to focus exclusively not just on the practicality of these tools, but also ways in which they could be used to further enhance the listening experience.
Thus, the art form was born. To this day, mastering remains a combination of practical and aesthetic processes. Though there isn’t any one ‘correct’ way to master, there are many recommended practices that mastering engineers follow.
Mastering advices for beginners from experts.
What’s your best advice for a beginner who is just starting out mastering, and wants to develop their skills?
- You have to start by listening: Listen to lots of very good recordings and become familiar with how they sound on the finest reproduction systems and compromised systems. Become familiar with the effects of PLR and PLR reduction and make sure you can identify when transients have been deleteriously affected (e.g. overcompression). Then try to obtain well-made raw mixes, which is the hard part. — Bob Katz, Digital Domain
- Spend all your initial efforts to create an accurate and high resolution monitor/room situation. That will enable you to refine your listening skills and eventually make good judgements on what may be needed. — Dave McNair, Dave McNair Mastering
- Do lots of ear training. EQ will be your number one tool, so get to know those key frequencies inside and out. — Ian Stewart, Ian Stewart Music
- Mastering is all about listening. The more variety of music you can listen to, the better foundation you will have. Listen with Purpose. Listen to the work of “the masters” and DO NOT get hung up on what type of gear they used or why it’s unfair that they got great mixes to work with. Study and learn – a lot – before you start turning knobs or clicking a mouse. — Scott Hull, Masterdisk
- Develop your skills on as many styles as you possibly can doing both recording and mixing for as long as you can before trying to make the jump to mastering. This is both art and science and besides technical and people skills you need to develop your ears, instincts and of course your own professional criteria but that can only be properly developed thru time. — Camilo Silva F., CamiloSilvaF.com
- Practice on a variety of local bands for no charge. Then listen to your results on a variety of real world playback systems. Find out what your monitors are doing to your masters. — Don Grossinger, DonGrossinger.com
- Pick a reference with an ideal 1) tonal balance, 2) density/punch, and 3) volume for your genre and tastes — and stick with it! Your job is to match your material to your reference in those three areas. Anything else is best addressed in the mix. — Brian Hazard, Resonance Mastering
- Mastering a song to match a reference song is like carving a block of wood to match a reference block of wood. Learn all the tools and techniques it takes to match a reference, and you’ll master anything for anyone (provided that the mix is decent and allows for that). — Janne Hatula, Fanu Music
- Embrace the mistakes you will inevitably make, and learn from them. Be gracious and generous to your clients whose music they entrust to you. — David Glasser, Airshow Mastering
- You can’t polish a turd… you need to learn how to mix well before your masters will start sounding good. Mastering can fix some mix errors, but if your mix has a lot of problems mastering tends to only exacerbate them with compression and limiting. My practical advice would be to try to master a new song everyday while watching YouTube tutorials and comparing your masters to professional tracks in your genre. If you keep this up long enough you’ll get very good eventually. — Zach Caraher, Big Z Mixing & Mastering Services
Producer Jon Griffin presents a straightforward guide to ensuring great results, and avoiding the many pitfalls, when mastering your own tracks.
Mastering: How To Master Your Tracks Like A Pro
Mastering is essentially the process of preparing your song, or collection of songs, for the commercial market. The aim of mastering is to present a coherent final product that translates well onto all kinds of listening systems and environments in the real world, beyond the relatively pristine confines of the studio.
In practice, mastering is primarily about fixing troublesome frequencies, lifting detail, balancing and enhancing the stereo image, and of course making the work competitive in terms of overall loudness. Of course, mastering can also involve more than this, but here we are going to focus on the essential processes that can be undertaken in your own studio, particularly when hiring a professional ME (mastering engineer) is not cost effective.
Your Pre-Mastered Mix
The more familiar you are with the mastering process, the more this can help you make good mixing decisions. Mix balance is king here, and so is maintaining headroom and a good dynamic range. Make sure none of your individual instruments or vocals go beyond 0dB where they will clip or distort: Even if your mix overall has good headroom and is well short of distorting, any peaks caused by tracks spiking above 0dB may become more apparent while mastering and severely compromise the mix.
Keep control of your mix dynamics by adding small doses of compression at different stages rather than heaping it on in one sitting, so a little compression while tracking, a little while mixing, a touch of limiting here and there and maybe even a touch on the mix buss itself. By the time you are printing off a mix, those compression touches will add up to a mix that is solid, without being lifeless and have just about the right headroom and dynamic range left that you or your ME would need.
You want to keep your loudest peaks with at least 1dB of headroom below zero, but really you can comfortably aim for greater margins, -3dB below zero would be even better. You don’t want to worry about ensuring your mix is loud – that is what mastering is for. Some engineers are even printing mixes at -18dB because they feel there is some sonic benefit. Your mix file can easily be brought up in level without issue with gain plugins or the clip gain functionality in most DAW’s. What you want to avoid at all costs are peaks above 0dB. It is far better to maintain headroom by printing a quieter mix than to squeeze every possible decibel out of it and risk going over before it even gets to mastering.
Dynamic range is the difference between the loudest and quietest moments in your music, and is also essential to preserve. A track with good dynamic range feels musical and exciting, whereas a track with poor dynamic range feels tight and fatiguing. How much dynamic range you build into any given mix is largely a judgement call you make based on taste, style and genre. Genres like pop and electronica tend to have less dynamic range than jazz, classical and other acoustic music. As a mix engineer you don’t necessarily want a mix that is too dynamic, but you certainly don’t want one that has no dynamics either! Meters like the Brainworx BX Meter that give real-time visual feedback on the dynamic range are popular tools and can help guide you in this respect.
Mastering Signal Chain
There are of course exceptions, and there are occasions where you have to do things differently, but the rule of thumb ME’s tend to agree on would be that a mastering chain should run something like this…
- Gain Plugins
- Stereo FX – such as widening or mastering reverb
You want to avoid using your limiter to deliver lots of gain at the end of your mastering chain. Ideally you only want to lean on them for a few dB, so make sure your audio file is at a good starting level either by using a gain plugin or, using the clip gain feature in most DAWS. Gaining will not affect your dynamic range only your headroom, you still want to keep enough headroom to apply your processes, but you don’t want the file to be so quiet that you are cranking the limiter to take up the slack.
Check a phase meter for good stereo representation: a nearly static line down the middle of the meter suggests there is little to no stereo quality; the result can be a lifeless, congested sounding mix. The solution could be as simple as inserting a basic widener and opening it up, fanning the mix out like a deck of cards. This leads to possibilities for additional surgical processes.
Well-known and highly experienced mastering engineer Craig Anderton preaches that EQ is 90% of the mastering process. If you are boosting or cutting EQ, a great piece of advice from Craig is to push the EQ frequency gain to where your ears want it…then halve your move. If you are boosting 3kHz by 3dB, bring it back to 1.5dB, that will probably be enough. In terms of EQing the mix generally, take time to listen first for the obvious things. Purposefully listen to the bass, the mid range, upper mid range and the highs. While trying to detect faults may seem like looking for a sonic needle in a haystack, start broad and you’ll gradually zero in on any issues if there are any. If you can’t detect anything you know you could improve, don’t EQ anything.
Stereo enhancement, or “widening”, involves spreading the various elements of a mix out over the stereo spectrum, pushing more sound to the extreme left and right.
This can be a significant and satisfying part of the mastering process, often transforming a track with a single turn of a knob. The downside is that it can also destroy a mix by either creating an un-real sense of space or by introducing phase issues and compromising energy levels.
The temptation can be to widen as far as your plugin will allow, but a more sensible approach is to apply it only to the point where you miss it when you take it out.
Widening to lift detail
Wideners can be especially useful for songs where a certain instrument or the vocal is getting lost in a busy mix. This is commonly because of an excess of information focused in the same ‘space’, either in terms of frequency or panning. Mixes that lack stereo information will be worse for that. Use a widener to fan out the mix, followed by an EQ boost to the fundamental frequency of the instrument or vocal you want more of. Typically you could look at boosting a vocal in the 3-4kHz range.
Compressors in Series
A useful technique to keep compression transparent and yet still achieve lots of gain is to use two in series, thus halving the workload on each. You get a cleaner, less obviously compressed sound because the circuits in each are being driven less and recovery times are near instantaneous.
The final stage is to cash in any remaining headroom and bring the mix level up as high as you can without clipping. Limiters are essentially compressors with super fast attack times and high compression ratios. You might start by setting the ceiling of a limiter to 0dB and then draw down the threshold to meet the audio peaks. The threshold is tied to an auto gain function, so the more you reduce the headroom and dig into the peaks, the more loudness you get back.
However, there is a trade-off here: the more you flatten the peaks, the less dynamic range you end up with. Mixes with too much limiting may appear loud, but in truth they feel flat, lacking dynamic energy and excitement. The trick is to be careful – a little limiting goes a long way, and heavy limiting very quickly gets ugly and amateurish.
One last point worth mentioning as a word of warning: you could hypothetically set your limiter to 0dB, thereby thoroughly exploiting any remaining headroom, and achieving maximum possible loudness. After all, you would think, if you have a limiter in place, you should be fine right? None shall pass and all that?
Well, yes, but there are certain digital processes that are required to smooth audio and in so doing they can add an additional thin layer of gain after the limiter: this could be enough to clip the master buss if your mix is already running right up to the limit. Therefore, it’s far better practice to allow perhaps 0.5db to act as a super safety net.